Merged revisions 322646 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r322646 | mnicholson | 2011-06-09 10:10:30 -0500 (Thu, 09 Jun 2011) | 5 lines

  don't drop any voice frames when checking for T.38 during early media

  (closes issue ASTERISK-17705)
  Review: https://reviewboard.asterisk.org/r/1186/
  patch by oej
  reported by oej
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@322668 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 1865c94a089252e456e8e891c7882299d48ae867..8a64834c1449d26229f957bc5c81a9e4d5fa8fae 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -6509,7 +6509,10 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
                }
                if (p) {
                        sip_pvt_lock(p);
-                       if (p->rtp) {
+                       if (p->t38.state == T38_ENABLED && !p->t38.direct) {
+                               /* drop frame, can't sent VOICE frames while in T.38 mode */
+                               break;
+                       } else if (p->rtp) {
                                /* If channel is not up, activate early media session */
                                if ((ast->_state != AST_STATE_UP) &&
                                    !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
@@ -6520,12 +6523,9 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
                                                transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
                                                ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
                                        }
-                               } else if (p->t38.state == T38_ENABLED) {
-                                       /* drop frame, can't sent VOICE frames while in T.38 mode */
-                               } else {
-                                       p->lastrtptx = time(NULL);
-                                       res = ast_rtp_write(p->rtp, frame);
                                }
+                               p->lastrtptx = time(NULL);
+                               res = ast_rtp_write(p->rtp, frame);
                        }
                        sip_pvt_unlock(p);
                }